view stream/stream_livedotcom.c @ 24589:9118be6575da

demux_audio.c: Fix timestamp handling The code calculated the pts values of audio packets by adding the length of the current packet to the pts of the previous one. The length of the previous packet should be added instead. This broke WAV timestamps near the end of the stream where a short packet occurs. Change the code to store the pts of the next packet instead of the last one. This fixes the WAV timestamps and allows some simplifications. MP3 timestamps are not affected as packets are always treated as constant decoded length, and FLAC timestamps still have worse problems (FLAC is treated as as if it was constant bitrate even though it isn't). Also store the timestamps as double instead of float.

#include "config.h"

#include <unistd.h>
#include <stdlib.h>
#include <stdio.h>
#include <string.h>

#include "stream.h"
#include "network.h"
#include "libmpdemux/demuxer.h"
#include "help_mp.h"

extern int network_bandwidth;

static int _rtsp_streaming_seek(int fd, off_t pos, streaming_ctrl_t* streaming_ctrl) {
  return -1; // For now, we don't handle RTSP stream seeking
}

static int rtsp_streaming_start(stream_t* stream) {
  stream->streaming_ctrl->streaming_seek = _rtsp_streaming_seek;
  return 0;
}


static int open_live_rtsp_sip(stream_t *stream,int mode, void* opts, int* file_format) {
  URL_t *url;

  stream->streaming_ctrl = streaming_ctrl_new();
  if( stream->streaming_ctrl==NULL ) {
    return STREAM_ERROR;
  }
  stream->streaming_ctrl->bandwidth = network_bandwidth;
  url = url_new(stream->url);
  stream->streaming_ctrl->url = check4proxies(url);
  //url_free(url);

  mp_msg(MSGT_OPEN, MSGL_INFO, "STREAM_LIVE555, URL: %s\n", stream->url);

  if(rtsp_streaming_start(stream) < 0) {
    mp_msg(MSGT_NETWORK,MSGL_ERR,"rtsp_streaming_start failed\n");
    goto fail;
  }

  *file_format = DEMUXER_TYPE_RTP;
  stream->type = STREAMTYPE_STREAM;
  return STREAM_OK;

fail:
  streaming_ctrl_free( stream->streaming_ctrl );
  stream->streaming_ctrl = NULL;
  return STREAM_ERROR;
}

static int open_live_sdp(stream_t *stream,int mode, void* opts, int* file_format) {
  int f;
  char *filename = stream->url;
  off_t len;
  char* sdpDescription;
  ssize_t numBytesRead;

  if(strncmp("sdp://",filename,6) == 0) {
    filename += 6;
#if defined(__CYGWIN__) || defined(__MINGW32__)
    f = open(filename,O_RDONLY|O_BINARY);
#else
    f = open(filename,O_RDONLY);
#endif
    if(f < 0) {
      mp_msg(MSGT_OPEN,MSGL_ERR,MSGTR_FileNotFound,filename);
      return STREAM_ERROR;
    }

    len=lseek(f,0,SEEK_END); 
    lseek(f,0,SEEK_SET);
    if(len == -1)
      return STREAM_ERROR;
    if(len > SIZE_MAX - 1)
      return STREAM_ERROR;

    sdpDescription = malloc(len+1);
    if(sdpDescription == NULL) return STREAM_ERROR;
    numBytesRead = read(f, sdpDescription, len);
    if(numBytesRead != len) {
      free(sdpDescription);
      return STREAM_ERROR;
    }
    sdpDescription[len] = '\0'; // to be safe
    stream->priv = sdpDescription;

    stream->type = STREAMTYPE_SDP;
    *file_format = DEMUXER_TYPE_RTP;
    return STREAM_OK;
  }
  return STREAM_UNSUPPORTED;
}


stream_info_t stream_info_rtsp_sip = {
  "standard RTSP and SIP",
  "RTSP and SIP",
  "Ross Finlayson",
  "Uses LIVE555 Streaming Media library.",
  open_live_rtsp_sip,
  {"rtsp", "sip", NULL },
  NULL,
  0 // Urls are an option string
};

stream_info_t stream_info_sdp = {
  "SDP stream descriptor",
  "SDP",
  "Ross Finlayson",
  "Uses LIVE555 Streaming Media library.",
  open_live_sdp,
  {"sdp", NULL },
  NULL,
  0 // Urls are an option string
};